In the past two main types of communication networks have evolved for transmitting information embedded in traffic streams: packet-oriented data networks and line-based voice networks. Their different quality-of-service (QoS) requirements are one aspect in which they differ from each other.
“Quality of Service” is defined differently depending on context and is therefore evaluated using different metrics. Known examples of metrics for measuring quality of service are the maximum number of information elements that can be transmitted (bandwidth), the number of information elements transmitted, the number of information elements not transmitted (loss rate), the—possibly mean—time delay during transmission ((transmission) delay), the—possibly mean—deviation from the otherwise standard interval between two information transmissions (delay jitter, interarrival jitter) or the number of information elements not permitted to be transmitted (blocking rate).
In multimedia networks services are also known as multimedia applications. A multimedia network is used here to describe a network in which a plurality of different services is provided. In a narrower sense it refers in particular to a broadband, service-integrated network (B-ISDN=Broadband Integrated Services Digital Network) in which the traffic streams resulting from use of the services can be transmitted by means of a standard, preferably packet-oriented transport mechanism. The term multimedia application thereby covers both services and normal telephony (also referred to as Voice over IP (VoIP) in packet-oriented IP networks, as well as services such as fax, telephone conference, video conference, Video on Demand (VoD) and so on.
Line-based (voice) networks are designed to transmit traffic streams in which continuously streaming (voice) information is embedded. In specialist circles these are also referred to as calls or sessions. Information is generally transmitted here with a high quality of service and security. For example for voice a minimum—e.g. <200 ms—delay is important without delay jitter, as voice requires a continuous information flow for playback in the receiving device. Information loss can therefore not be compensated for by retransmission of information not transmitted and generally results in the receiving device in an acoustically perceptible clicking. In specialist circles voice transmission is also generally referred to as a realtime (transmission) service.
A low blocking rate is achieved for example by appropriate dimensioning and planning of the voice networks. A small and largely constant delay or delay jitter is generally also achieved in the case of joint transmission of a plurality of traffic streams via a shared channel by using a static time division multiplex also referred to as TDM. Here the traffic streams are segmented in the transmitter into homogenous units of fixed length—also referred to as time slots—and transmitted temporally interleaved in each other. Assignment of the time slots to the respective traffic streams is indicated by their position within the channel. After joint transmission the time slots can be assigned to their associated traffic streams in the receiver and where necessary can also be reassembled into the original traffic streams. As a result the transmission capacity of the traffic streams is essentially not subject to any fluctuations during line-based transmission but is fixed at a predefined value (e.g. 64 kbps in modern ISDN telephone networks).
Packet-oriented (data) networks are designed to transmit traffic streams configured as packet streams, also referred to in specialist circles as data packet streams. It is generally not necessary to guarantee a high quality of service here. For example in the case of an email it is not necessary to have a minimum delay without delay jitter, as an email does not have to be played back in realtime at the receiver. More important here is that the email should be transmitted without error. Information loss is therefore generally compensated for by retransmission of information that was not transmitted or was transmitted incorrectly. The delay of an email therefore varies as a function of the frequency of retransmission. Delay jitter therefore also tends to be high. In specialist circles the transmission of data is therefore also referred to as a non-realtime service.
There is essentially no blocking rate in packet-oriented data networks. In principle all packets in all traffic streams are always transmitted. The traffic streams are however transmitted even when there is only moderate loading of a data network with significantly fluctuating time delays as the individual packets are generally transmitted in the sequence of their network access, i.e. the time delays increase, the more packets have to be transmitted by a data network. Joint transmission of a plurality of traffic streams via a shared channel is generally achieved by using a statistical (time division) multiplex. Here the packets in the traffic streams in the transmitter are transmitted interleaved in time according to statistical rules. The rules could for example specify that the packets are to be transmitted in the sequence of their arrival (best effort). If a plurality of packets arrives at the same time, one is transmitted while the remainder are temporarily buffered, resulting in an increase in delay jitter. If more packets than can be buffered arrive at the same time, the surplus packets are discarded. Assignment of the packets to the respective traffic streams is indicated by assignment information in the packet overhead (comprising a header and/or trailer). After joint transmission therefore the packets can be assigned to their associated traffic streams in the receiver. The transmission capacity of the traffic streams is essentially not subject to limitations during packet-oriented transmission but can in principle (in the context of the capacity of the shared channel) have a different value at any time.
In the course of the convergence of line-based voice and packet-oriented data networks, voice transmission services and in future also faster broadband services such as for example the transmission of moving image information (VoD, video conference) will be provided in service-integrated packet-oriented (multimedia) networks—also referred to as voice/data networks, i.e. realtime services until now generally transmitted in a line-based manner are now transmitted in packet streams in a convergent voice/data network. These are also referred to as realtime packet streams. This gives rise to the problem that a high quality of service and security are necessary for the packet-oriented provision of a realtime service to ensure that this is comparable in quality to a line-based transmission, while modern (packet-oriented) data networks and in particular the internet have no adequate mechanisms to guarantee a high quality of service.
Quality of service requirements in service-integrated, packet-oriented networks generally apply to all network types. They are independent of the specific configuration of the packet orientation. The packets can therefore be configured as internet, X.25 or frame relay packets and also as ATM cells. They are sometimes also referred to as messages, particularly when a message is transmitted in a packet. Data packet streams and realtime packet streams are hereby exemplary embodiments of traffic streams transmitted in communication networks. Traffic streams are also referred to as connections, even in packet-oriented networks, in which connectionless transmission technology is deployed. For example information is transmitted with TCP/IP using what are known as flows, by means of which, despite the connectionless nature of IP, transmitter and receiver (e.g. web server and browser) are connected at a logically abstract level, i.e. in a logically abstract way flows also represent connections. It is only essential for a connection that a connection setup takes place before transmission, during which process a context is created which continues to exist at least during transmission. An explicit clear down of the connection can take place after transmission. Implicit mechanisms such as for example timeout of the connection after a specified transmission-free period are however also possible.
The best known data network at present is the internet. The internet is conceived as an open (long-range) data network with open interfaces to connect (usually local and regional) data networks of different manufacturers. The main focus to date has therefore been on the provision of a manufacturer-independent transport platform. Adequate mechanisms for guaranteeing quality of service play a secondary role and therefore barely exist.
The convergence of telecommunication (also known as voice networks) and the conventional data world (also known as data networks) into IP (internet protocol) based networks and services is a difficult task in respect of IP technology, as this is designed as a packet-oriented data network primarily for “best effort” transmission and at best provides for compliance with rather vaguely formulated service level agreements (SLA), while in the case of telecommunication very stringent requirements relating to QoS, reliability, availability and security of network and services play a major role. The internet world responds to this task with a plurality of increasingly complex and expensive solutions but has not as yet found a total solution that is also manageable and workable from an economic point of view.
The QoS requirements of a service or an application in respect of a network can be defined using different criteria, of which some examples are given below:
the throughput characteristics of the digitally coded information, i.e. the necessary bandwidth or bandwidth characteristics (fixed bandwidth, variable bandwidth [e.g. with mean value, peak value, ‘burstiness’ factor or other characterizing parameters]) and susceptibility to information losses,
the delay characteristics, i.e. the effects of an absolute delay (transit time from information source to information sink) and susceptibility to runtime fluctuations or delay jitter (of course delay jitter can be converted to absolute delay by buffering but this is usually very complex),
the necessary or unnecessary temporal consistency or time invariance of the transmitted information, i.e. whether the information units have to be delivered in exactly the same sequence in which they arrived or not (in some cases the compatibility or incompatibility of higher service and application layers must also be taken into account).
The consequences of different QoS requirements can be clarified using two examples:
Unidirectional audio/video applications (e.g. streaming video) require realtime presentation at the receiver but in most cases it is immaterial whether the absolute delay is 1/100, 1 or 5 seconds, as long as there is continuity after the start of playback. Such delay tolerance could for example be used to compensate for information losses using repeats, thereby improving the quality. Alternatively transmission could also take place with redundancy (higher bandwidth) to compensate for possible data losses.
Interactive, i.e. bidirectional realtime communication (voice, video, etc.) between people must take into account the response capability and typical communication and dialog behaviors of people. Here the absolute delay (and therefore of course also the delay jitter) must be limited to a few hundred milliseconds (e.g. 200 ms). On the other hand in some instances somewhat higher loss rates can be tolerated, as the capacity of the human brain to “smooth out irregularities” in speech and visual perception is very well-developed and alertness to minor defects is somewhat reduced in dialog. Realtime dialogs between machines are more complex, however. In this case it may be that attention must be focused on the completeness of the information and on short delays close to the physical limit due to geographical distance (transit time approx. 5 ms per 1000 km distance).
If the QoS specifications are defined and if a network still has reserves in one of these areas, it can deploy these to compensate for deficits in another area. Such compensation can be clarified using two examples:
If an application tolerates relatively high information losses, the delay jitter can be reduced by discarding information units which have been subjected to a high level of delay. Conversely larger delay jitter can of course also be deployed to achieve lower losses, which however results in large buffers.
If the maximum for delay jitter is below the minimum time interval of the incoming information units (known as a ‘fast network’), there are no problems with the temporal consistency of the transmitted information. If measures are provided to restore this temporal consistency, relatively large delay jitter can be tolerated as long as the framework of the absolute permissible delay is not exceeded.
As well as QoS the general availability of services is also an important parameter that depends to a large degree on the network and its characteristics. In the event of an error, e.g. in the case of failure of individual network components or connecting lines, is a backup path available and how quickly can it be brought into use? Do interruptions occur that the user can identify and how long do these last? Does the network operator or even the user have to intervene in any way to restore the service in some instances? The reliability of the network in itself and the way in which it can help to bypass errors and where necessary restore the applications is of great significance here.
A standard network must therefore be considered subject to qualification by initial conditions as proposed here and of course it should also be achieved in the most efficient manner possible, i.e. at the lowest possible cost and in an economically advantageous manner.
The known network technologies satisfy the above specification partially at best.
The simplest approach is the tried and tested technology of circuit switching, with which a dedicated connection (in the bidirectional instance or with multiple relations where necessary also two or more connections) (sometimes also referred to casually as a path) with a permanently assigned and absolutely reserved bandwidth is switched for every communication relation. Such connections are either configured explicitly as individual physical lines (e.g. copper wires) or as (virtual) channels in what are known as transmission or switching systems, which allow multiple utilization of physical lines. A mix with differently implemented links is also possible. The possible data throughput of such a connection is determined by its own or its assigned bandwidth, the transport delay time is made up of the propagation delay, i.e. the distance-dependent transit time on the line, and the switching delays, i.e. the inherent processing times resulting during switching of the digitally coded information (data) in the network nodes (switches). Switching here means transferring information (data) from a defined incoming line/channel to an outgoing line/channel specified when the connection is being set up. Both delay components can generally (i.e. when the systems are operating without interference) be assumed to be constant for the period of a communication relation (with through-connected path or existing connection). When there is no interference therefore the same quasi-optimum QoS is predefined and achievable for all applications (no information losses, constant, generally relatively short, delay, no transpositions). However for this the connection must be permanently switched (and reserved) for the duration of the communication relation, even if the application only uses it very infrequently (e.g. only sporadically). Reliability/availability can be improved by switching as quickly as possible to a previously provided alternative connection in the event of an error (double capacity required) or switching the backup connection immediately (delay and expense, particularly when a plurality of connections is affected at the same time by one failure).
Packet switching technology aims at better utilization of resources (bandwidth) by flexible sharing of lines and (where necessary virtual) channels or switching and transmission media by a plurality of communication relations. Known, modern representatives are for example the connection-oriented ATM technology with fixed-length packets (also referred to as cells) and the connectionless IP technology with variable length packets.
ATM technology is also promoted at the ITU-T under this name and with the objective of broadband ISDN (B-ISDN). ATM has mechanisms to provide a broad spectrum of service classes with defined and guaranteed QoS (at the statistical mean), even with very scant resources (available bandwidths). The resulting systems and networks are therefore very complex and expensive. Dimensioning and operation require highly qualified specialist personnel. ATM operates in a connection-oriented manner, with a network of ‘virtual’ paths and channels, assigned to each other in a hierarchical manner. For a plurality of different service classes bandwidths can be reserved in a connection-specific manner and also ‘guaranteed’ based on the traffic statistics used as a basis. Different queuing and scheduling mechanisms are used for this and these can be set in every node for each path and channel (connection) by means of appropriate parameters. Fine-granular dimensioning and connection acceptance requirements can be used to limit information losses and the variable parts switching delays (these are essentially determined by queuing) based on statistical rules. Owing to the connection-oriented mode of operation, transposition of information units is unlikely during interference-free operation. As a result of the connection orientation all inherent mechanisms have to be executed again during error handling. The basic concepts are therefore often very similar to those of circuit switching technology.
IP technology is more of a pragmatic approach that has become established in the data world due to its simple basic mechanisms. It has made massive progress in recent years so that the capacity (data throughput, control efficiency) of systems and networks based on it is comparable to that of systems based on ATM technology. The success of IP technology is significantly due to the fact that a large part of the services and applications are already based on packet-oriented internet protocols (IP) in the terminal. It is currently predicted that the growth in IP-based services will also be significantly greater in the future than in other technologies, so an extensive migration of all services to transport via IP-based networks seems probable. Unlike ATM networks IP networks operate in a connectionless manner and only provide a ‘best effort’ service, with which it is difficult to predict and impossible to guarantee an achievable QoS even with generously dimensioned networks.
The following solutions were also known to date:
a) Using an ATM network as a core network. Edge devices transfer the IP data streams to ATM connections of appropriate service classes and transport takes place in corresponding connections in the ATM network. Problems here are scalability, complexity and setting up and operating costs (see ATM technology above). This solution is of more assistance in the core. The same disadvantages apply to (additional) use in the access. The following solution is an alternative in the access.b) Using a signaling protocol and setting up connections with reserved bandwidths via the IP network (integrated services—IntServ, RSVP). This solution is feasible in principle both end-to-end (E2E), i.e. from terminal to terminal, and on subsections. It can be used for each communication flow or (in the core) also for aggregated communication flows. It is however elaborate, expensive, non-scaling (control costs) and inefficient, i.e. very similar to ATM technology.c) MPLS: This approach is based on ATM technology. Paths (connections) are set up in the network, via which the traffic of individual (generally aggregated) flows is specifically routed. It is frequently proposed for QoS in conjunction with RSVP and DiffServ (see below under d)) and can also be provided based on ATM transport. It reverts to the complexity of connection-oriented mechanisms with all the consequences already set out (from bandwidth control to monitoring the existence of the connection), i.e. it is of similar complexity to ATM technology. In conjunction with the DiffServ solution it should in particular alleviate the problem discussed there (specific traffic control via paths).d) Differentiated Services (DiffServ): The data packets are classified and marked in the edge device on the basis of their association with specific services, applications or communication relations, etc. (Flow-related) access control and monitoring (e.g. for availability of resources and compliance with the specified bandwidth and QoS characteristics) can and should also take place. The packets then follow the route through the network predefined by their packet header information (e.g. destination address) and the routing protocols, whereby they are processed (or prioritized) in every node according to their marking with appropriate ‘per hop’ behavior. The DiffServ approach allows the freedom of per hop behavior within a single routing domain, e.g. the (sub) network of an operator, but requires complete edge processing between such domains (subnetworks). The DiffServ approach cannot prevent temporary and/or local bottlenecks, as there is generally no consideration of or harmonization with the routes predefined by the routing protocols. Generally packets with the same destination follow the same set route from the point when they meet in a node. This can result significantly in skewed loads and bottlenecks in the networks with correspondingly long (queuing) delays or even packet losses. Network and route engineering is also a complex task, whereby the aspects of reliability and availability (e.g. rerouting in the event of error) are a further complication.
In principle almost all combinations of said approaches are conceivable and have to a large extent also been discussed. All these approaches have in common the fact that (with the exception of DiffServ) they are based on paths and use bandwidths and where necessary further resources reserved along said paths. Even a purely DiffServ approach is always based at least on routes predefined by routing protocols. This is generally associated with a major administrative burden with regard to preparing and (statically) setting up paths and routes in the network or a correspondingly high control burden for the dynamic selection and switching of the routes. Also storage devices must be kept available in every network node to hold path-specific and connection-specific information, which can be lost or have to be reconfigured on other routes in the event of error. Even with the purely DiffServ approach the traffic follows the routes predefined by the routing protocols and these therefore have to be very carefully dimensioned and monitored. Generally however it is not possible to predict exactly either all fluctuations in traffic volume or the responses of the routing protocols to possible events in the network.